Voice-over-IP (VoIP) has been steadily gaining in popularity within the personal computing and carrier communities. As technology advances, running similar services in cellular devices becomes viable. A key technical question is how to adapt the services to cope with reduced bandwidth and significantly less processing power compared to a desktop PC.
Push-to-talk-over-cellular (PoC) defines a half-duplex VoIP system for mobile devices. By using the packet-switched capabilities of wireless data networks the service is not restricted geographically—unlike conventional two-way radio systems such as private mobile radio (PMR).
To make PoC a success, handset and network performance should be optimal. Any performance impacting delays could mean the difference between the success and failure of the service.
The audio quality of PoC is limited by the available GPRS/EGPRS bandwidth on the cellular system. Audio data is transmitted across a PoC system using Adaptive Multi-Rate (AMR) coding packaged in the Real-Time Transport Protocol (RTP) on top of the User Datagram Protocol (UDP) unacknowledged transport protocol. As a result lost packets are never retransmitted.
PTT is a real time application where buffering of audio data is kept to a minimum to reduce latency from the beginning of a talk burst to the beginning of playback. Each RTP packet is time stamped to allow the system to discard delayed packets, which in turn prevents the talk burst from growing unbounded.
Currently, techniques have been derived to improve downlink streaming of AMR RTP packets by buffering to reduce jitter and packet re-ordering to correct packets received out of order. However, it has been shown that limited bandwidth in the uplink (send side) causes lost packets which can not be corrected in the downlink. This bandwidth limitation can be caused by congestion, or by systems with minimal data resources. Currently, the only mechanism to reduce bandwidth in the uplink is to change AMR coding rates. This approach alone does not provide enough reduction in bandwidth to prevent lost packets due to minimum bandwidth on the GPRS link. Problems due to limited bandwidth have been observed while using the lowest 4.75 kb AMR coding rate.
What is needed is a system and/or method to further conserve uplink bandwidth in a PoC or other conversational service IP based application using an AMR payload format.